Fix the different samples per channel in aecdump

BUG=webrtc:3359
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6996 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
aluebs@webrtc.org 2014-08-28 10:43:09 +00:00
parent deaece6ac0
commit 59a1b1b928

View File

@ -367,7 +367,8 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
if (debug_file_->Open()) {
event_msg_->set_type(audioproc::Event::STREAM);
audioproc::Stream* msg = event_msg_->mutable_stream();
const size_t channel_size = sizeof(float) * samples_per_channel;
const size_t channel_size =
sizeof(float) * fwd_in_format_.samples_per_channel();
for (int i = 0; i < fwd_in_format_.num_channels(); ++i)
msg->add_input_channel(src[i], channel_size);
}
@ -384,7 +385,8 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
audioproc::Stream* msg = event_msg_->mutable_stream();
const size_t channel_size = sizeof(float) * samples_per_channel;
const size_t channel_size =
sizeof(float) * fwd_out_format_.samples_per_channel();
for (int i = 0; i < fwd_proc_format_.num_channels(); ++i)
msg->add_output_channel(dest[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile());
@ -533,7 +535,8 @@ int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
if (debug_file_->Open()) {
event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
const size_t channel_size = sizeof(float) * samples_per_channel;
const size_t channel_size =
sizeof(float) * rev_in_format_.samples_per_channel();
for (int i = 0; i < num_channels; ++i)
msg->add_channel(data[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile());